vicidial.org

by **gardo** » Thu Jan 22, 2009 9:37 am

Hi Matt,

I have noticed with survey dialing (where other end presses "1" to get to an agent and "8" to be included in the DNC), that with SIP accounts, the DTMF tones don't get detected properly. I have tried it with different SIP providers using ulaw and g729. However, I get the same results. I have to continuously press "1" or "8" get the desired extension. Putting VoIP phone numbers and cellphone numbers makes it almost impossible to finish the survey script. Mostly nothing happens when pressing any of the options.

I'm using the following:

astguiclient 2.0.4.1rc3
asterisk 1.2.27

I haven't upgraded the system to 2.0.4.1rc5 yet due to some customizations.

by **Op3r** » Thu Jan 22, 2009 7:17 pm

have you check your dtmfmode at the sip.conf?

use rfc2833

by **mflorell** » Fri Jan 23, 2009 6:31 pm

What is the dtmf_mode?

by **gardo** » Sat Jan 24, 2009 2:02 am

I tried both inband and rfc2833. I've had some success with inband. However, it's still unreliable. Out of more than 15 calls, I can only make 1 call pass DTMF.

by **mflorell** » Sat Jan 24, 2009 7:48 pm

I'm not really familiar with this problem on SIP, it's usually not an issue. On T1s I would check the relaxdtmf setting.

Have you tried turning on sip debug and seeing what shows up during this?(only one call at a time of course off hours)

by **gardo** » Sun Jan 25, 2009 8:23 am

Here's the scenario:

Dialing normally and reaching an IVR, DTMF tones work properly (without VICIDIAL) over SIP.

Now using the the broadcast and survey scripts, when other end answers the line and presses any of the given options (press 1, 2, 8 and etc), DTMF tones don't get detected.

by **mflorell** » Sun Jan 25, 2009 8:37 am

Does this happen over IAX as well connecting to another Asterisk server?

What is the loadavg when this happens?

Can this happen with only one call going on?

by **gardo** » Sun Jan 25, 2009 4:18 pm

Load average is pretty much negligible since I'm using one line for debugging purposes.

Yes it's happenning on just 1 call. I'm doing it one call at a time.

I have enabled the logging options for VICIDIAL and checked the logs. Unfortunately, it's not showing me anything related to DTMF not being detected properly.

I haven't tried IAX yet since I don't have an IAX provider.

by **williamconley** » Mon Jan 26, 2009 12:08 am

I bumped into this same issue and traced it to the provider in question. I changed providers and found that suddenly everything was working. I have yet to complete the troubleshooting with the provider in question, as the resolution was simple enough to just change providers.

Strangely enough, it happened overnight. It worked ... then suddenly it stopped working. And this provider is both IAX and SIP.

Open a test account with another provider (such as, perhaps, one recommended by vicidial ... binfone?). If that fixes your problem, then go back to your provider and describe the problem a little more thoroughly ("You are about to lose a customer" is an apt description).

If not, have a look at your "SIP-INVITE" and see if it contains the appropriate dtmf method. sip debug should show it for you.

by **gardo** » Wed Jan 28, 2009 2:36 am

Did some troubleshooting with the provider and we were able to make it work using G711 as codec. However, using G729 wasn't a success. They haven't send me the details on the configuration changes they made to make it work.

Williamconley, what codecs are you using with your provider that is working with the survey scripts?

by **mcargile** » Wed Jan 28, 2009 12:23 pm

With any of the compressed codecs GSM, G729, etc you have to have dtmfmode set to rfc2833. Inband does not work with compression.

Are you seeing any NOTICES, WARNINGS, or ERRORS in the asterisk CLI during this conversation?

by **gardo** » Wed Jan 28, 2009 6:40 pm

I agree with you since those compressed codecs don't really support inband. DTMF mode has been set to rfc2833.

Ran a couple of tests with the provider monitoring using ulaw/g711 as codec. They figured that it was an issue on their end. They got it working with g711 but not with g729.

There are actually no warnings or error messages that I can see with SIP debug.

We're trying out another provider and see if it works with them.

by **williamconley** » Fri Jan 30, 2009 12:31 am

i tried with ulaw, gsm & g729 IAX & SIP before changing providers.

by **gardo** » Fri Jan 30, 2009 12:58 pm

Changing providers seem to solve the issue. Works with g729 and g711.

by **williamconley** » Fri Jan 30, 2009 4:42 pm

now get your sip-invite (from BOTH providers) and go back to the original provider and ask why duplicate sip-invites result in different call capabilities. They may resolve the issue so you have two providers again.

by **tsvision** » Fri Feb 13, 2009 9:22 pm

change your DTMF=auto in your sip.conf

by **kolucoms6** » Thu Feb 19, 2009 3:55 pm

Tried DTMF=auto but doesnt detect DTMF

by **williamconley** » Thu Feb 19, 2009 4:59 pm

dtmfmode (not dtmf)

by **kolucoms6** » Thu Feb 19, 2009 5:01 pm

ooppss Yes

Type error :-)

dtmfmode=auto

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survey dialing unable to detect dtmf tones

survey dialing unable to detect dtmf tones

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